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Imagine I have H.264 AnxB frames coming in from a real-time conversation. What is the best way to encapsulate in MPEG2 transport stream while maintaining the timing information for subsequent playback?

I am using libavcodec and libavformat libraries. When I obtain pointer to object (*pcc) of type AVCodecContext, I set the foll.

pcc->codec_id = CODEC_ID_H264;
pcc->bit_rate = br;
pcc->width = 640;
pcc->height = 480;
pcc->time_base.num = 1;
pcc->time_base.den = fps;

When I receive NAL units, I create a AVPacket and call av_interleaved_write_frame().

AVPacket pkt;
av_init_packet( &pkt );
pkt.flags |= AV_PKT_FLAG_KEY;   
pkt.stream_index = pst->index;
pkt.data = (uint8_t*)p_NALunit;
pkt.size = len;

pkt.dts = AV_NOPTS_VALUE;
pkt.pts = AV_NOPTS_VALUE;

av_interleaved_write_frame( fc, &pkt );

I basically have two questions:

1) For variable framerate, is there a way to not specify the foll. pcc->time_base.num = 1; pcc->time_base.den = fps; and replace it with something to indicate variable framerate?

2) While submitting packets, what "timestamps" should I assign to pkt.dts and pkt.pts?

Right now, when I play the output using ffplay it is playing at constant framerate (fps) which I use in the above code.

I also would love to know how to accommodate varying spatial resolution. In the stream that I receive, each keyframe is preceded by SPS and PPS. I know whenever the spatial resolution changes. IS there a way to not have to specify pcc->width = 640; pcc->height = 480; upfront? In other words, indicate that the spatial resolution can change mid-stream.

Thanks a lot, Eddie

user1058600
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1 Answers1

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DTS and PTS are measured in a 90 KHz clock. See ISO 13818 part 1 section 2.4.3.6 way down below the syntax table.

As for the variable frame rate, your framework may or may not have a way to generate this (vui_parameters.fixed_frame_rate_flag=0). Whether the playback software handles it is an ENTIRELY different question. Most players assume a fixed frame rate regardless of PTS or DTS. mplayer can't even compute the frame rate correctly for a fixed-rate transport stream generated by ffmpeg.

I think if you're going to change the resolution you need to end the stream (nal_unit_type 10 or 11) and start a new sequence. It can be in the same transport stream (assuming your client's not too simple).

Mutant Bob
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